Commit from LuCI Translation Portal by user jow.: 105 of 105 messages translated...
[project/luci.git] / po / zh_CN / pbx.po
1 msgid ""
2 msgstr ""
3 "Project-Id-Version: PACKAGE VERSION\n"
4 "Last-Translator: Automatically generated\n"
5 "Language-Team: none\n"
6 "MIME-Version: 1.0\n"
7 "Content-Type: text/plain; charset=UTF-8\n"
8 "Content-Transfer-Encoding: 8bit\n"
9
10 msgid "Advanced Settings"
11 msgstr ""
12
13 msgid "Available"
14 msgstr ""
15
16 msgid ""
17 "Avoid using anything but alpha-numeric characters, space, comma, and period."
18 msgstr ""
19
20 msgid "Away"
21 msgstr ""
22
23 msgid "Blacklisted Numbers"
24 msgstr ""
25
26 msgid "Call Routing"
27 msgstr ""
28
29 msgid "Call-through Numbers"
30 msgstr ""
31
32 msgid "Copy-paste large lists of numbers here."
33 msgstr ""
34
35 msgid ""
36 "Designate numbers that are allowed to call through this system and which "
37 "user's privileges it will have."
38 msgstr ""
39
40 msgid "Dials numbers unmatched elsewhere"
41 msgstr ""
42
43 msgid "Do Not Disturb"
44 msgstr ""
45
46 msgid "Domain/IP Address/Dynamic Domain"
47 msgstr ""
48
49 msgid "Dynamic List of Blacklisted Numbers"
50 msgstr ""
51
52 msgid "Email"
53 msgstr ""
54
55 msgid "Enable Incoming Calls (Register via SIP)"
56 msgstr ""
57
58 msgid "Enable Incoming Calls (set Status below)"
59 msgstr ""
60
61 msgid "Enable Outgoing Calls"
62 msgstr ""
63
64 msgid "Enabled"
65 msgstr ""
66
67 msgid ""
68 "Enter phone numbers that you want to decline calls from automatically. You "
69 "should probably omit the country code and any leading zeroes, but please "
70 "experiment to make sure you are blocking numbers from your desired area "
71 "successfully."
72 msgstr ""
73
74 msgid ""
75 "Enter this IP (or IP:port) in the Server/Registrar setting of SIP devices "
76 "you will use ONLY locally and never from a remote location."
77 msgstr ""
78
79 msgid ""
80 "Enter this hostname (or hostname:port) in the Server/Registrar setting of "
81 "SIP devices you will use from a remote location (they will work locally too)."
82 msgstr ""
83
84 msgid "External SIP Port"
85 msgstr ""
86
87 msgid ""
88 "For each provider enabled for incoming calls, here you can restrict which "
89 "users to ring on incoming calls. If the list is empty, the system will "
90 "indicate that all users enabled for incoming calls will ring. Invalid "
91 "usernames will be rejected silently. Also, entering a username here "
92 "overrides the user's setting to not receive incoming calls. This way, you "
93 "can make certain users ring only for specific providers. Entries can be made "
94 "in a space-separated list, and/or one per line by hitting enter after every "
95 "one."
96 msgstr ""
97
98 msgid ""
99 "For each user enabled for outgoing calls you can restrict what providers the "
100 "user can use for outgoing calls. By default all users can use all providers. "
101 "To show up in the list below the user should be allowed to make outgoing "
102 "calls in the \"User Accounts\" page. Enter VoIP providers in the format "
103 "username@some.host.name, as listed in \"Outgoing Calls\" above. It's easiest "
104 "to copy and paste the providers from above. Invalid entries, including "
105 "providers not enabled for outgoing calls, will be rejected silently. Entries "
106 "can be made in a space-separated list, and/or one per line by hitting enter "
107 "after every one."
108 msgstr ""
109
110 msgid "Full Name"
111 msgstr ""
112
113 msgid "General Settings"
114 msgstr ""
115
116 msgid "Google Accounts"
117 msgstr ""
118
119 msgid "Google Talk Status"
120 msgstr ""
121
122 msgid "Google Talk Status Message"
123 msgstr ""
124
125 msgid "Google Voice/Talk Accounts"
126 msgstr ""
127
128 msgid ""
129 "Here you must configure at least one SIP account, that you will use to "
130 "register with this service. Use this account either in an Analog Telephony "
131 "Adapter (ATA), or in a SIP software like CSipSimple, Linphone, or Sipdroid "
132 "on your smartphone, or Ekiga, Linphone, or X-Lite on your computer. By "
133 "default, all SIP accounts will ring simultaneously if a call is made to one "
134 "of your VoIP provider accounts or GV numbers."
135 msgstr ""
136
137 msgid ""
138 "If setting Server/Registrar to %s or %s does not work for you, try setting "
139 "it to %s or %s and entering this port number in a separate field that "
140 "specifies the Server/Registrar port number. Beware that some devices have a "
141 "confusing setting that sets the port where SIP requests originate from on "
142 "the SIP device itself (the bind port). The port specified on this page is "
143 "NOT this bind port but the port this service listens on."
144 msgstr ""
145
146 msgid ""
147 "If you experience jittery or high latency audio during heavy downloads, you "
148 "may want to enable QoS. QoS prioritizes traffic to and from your network for "
149 "specified ports and IP addresses, resulting in better latency and throughput "
150 "for sound in our case. If enabled below, a QoS rule for this service will be "
151 "configured by the PBX automatically, but you must visit the QoS "
152 "configuration page (Network->QoS) to configure other critical QoS settings "
153 "like Download and Upload speed."
154 msgstr ""
155
156 msgid ""
157 "If you have more than one account that can make outgoing calls, you should "
158 "enter a list of phone numbers and/or prefixes in the following fields for "
159 "each provider listed. Invalid prefixes are removed silently, and only 0-9, "
160 "X, Z, N, #, *, and + are valid characters. The letter X matches 0-9, Z "
161 "matches 1-9, and N matches 2-9. For example to make calls to Germany through "
162 "a provider, you can enter 49. To make calls to North America, you can enter "
163 "1NXXNXXXXXX. If one of your providers can make \"local\" calls to an area "
164 "code like New York's 646, you can enter 646NXXXXXX for that provider. You "
165 "should leave one account with an empty list to make calls with it by "
166 "default, if no other provider's prefixes match. The system will "
167 "automatically replace an empty list with a message that the provider dials "
168 "all numbers not matched by another provider's prefixes. Be as specific as "
169 "possible (i.e. 1NXXNXXXXXX is better than 1). Please note all international "
170 "dial codes are discarded (e.g. 00, 011, 010, 0011). Entries can be made in a "
171 "space-separated list, and/or one per line by hitting enter after every one."
172 msgstr ""
173
174 msgid "Incoming Calls"
175 msgstr ""
176
177 msgid "Insert QoS Rules"
178 msgstr ""
179
180 msgid "Makes Outgoing Calls"
181 msgstr ""
182
183 msgid "NOTE: There are no Google or SIP provider accounts configured."
184 msgstr ""
185
186 msgid ""
187 "NOTE: There are no Google or SIP provider accounts enabled for incoming "
188 "calls."
189 msgstr ""
190
191 msgid ""
192 "NOTE: There are no Google or SIP provider accounts enabled for outgoing "
193 "calls."
194 msgstr ""
195
196 msgid "NOTE: There are no local user accounts configured."
197 msgstr ""
198
199 msgid "NOTE: There are no local user accounts enabled for outgoing calls."
200 msgstr ""
201
202 msgid "No"
203 msgstr ""
204
205 msgid "Number of Seconds to Ring"
206 msgstr ""
207
208 msgid "Outbound Proxy"
209 msgstr ""
210
211 msgid "Outgoing Calls"
212 msgstr ""
213
214 msgid "PBX Main Page"
215 msgstr ""
216
217 msgid "PBX Service Status"
218 msgstr ""
219
220 msgid "PIN"
221 msgstr ""
222
223 msgid "Password"
224 msgstr ""
225
226 msgid ""
227 "Pick a random port number between 6500 and 9500 for the service to listen "
228 "on. Do not pick the standard 5060, because it is often subject to brute-"
229 "force attacks. When finished, (1) click \"Save and Apply\", and (2) click "
230 "the \"Restart VoIP Service\" button above. Finally, (3) look in the \"SIP "
231 "Device/Softphone Accounts\" section for updated Server and Port settings for "
232 "your SIP Devices/Softphones."
233 msgstr ""
234
235 msgid "Port Setting for SIP Devices"
236 msgstr ""
237
238 msgid "Providers Used for Outgoing Calls"
239 msgstr ""
240
241 msgid "QoS Settings"
242 msgstr ""
243
244 msgid "RTP Port Range End"
245 msgstr ""
246
247 msgid "RTP Port Range Start"
248 msgstr ""
249
250 msgid ""
251 "RTP traffic carries actual voice packets. This is the start of the port "
252 "range that will be used for setting up RTP communication. It's usually OK to "
253 "leave this at the default value."
254 msgstr ""
255
256 msgid "Receives Incoming Calls"
257 msgstr ""
258
259 msgid "Remote Usage"
260 msgstr ""
261
262 msgid "Rings users enabled for incoming calls"
263 msgstr ""
264
265 msgid "SIP Accounts"
266 msgstr ""
267
268 msgid "SIP Device/Softphone Accounts"
269 msgstr ""
270
271 msgid "SIP Provider Accounts"
272 msgstr ""
273
274 msgid "SIP Realm (needed by some providers)"
275 msgstr ""
276
277 msgid "SIP Server/Registrar"
278 msgstr ""
279
280 msgid "SIP Server/Registrar Port"
281 msgstr ""
282
283 msgid "Server Setting"
284 msgstr ""
285
286 msgid "Server Setting for Local SIP Devices"
287 msgstr ""
288
289 msgid "Server Setting for Remote SIP Devices"
290 msgstr ""
291
292 msgid "Service Status"
293 msgstr ""
294
295 msgid ""
296 "Set the number of seconds to ring users upon incoming calls before hanging "
297 "up or going to voicemail, if the voicemail is installed and enabled."
298 msgstr ""
299
300 msgid "Space-Separated List of Blacklisted Numbers"
301 msgstr ""
302
303 msgid "Specify numbers individually here. Press enter to add more numbers."
304 msgstr ""
305
306 msgid ""
307 "The number(s) specified above will be able to dial out with this user's "
308 "providers. Invalid usernames, including users not enabled for outgoing "
309 "calls, are dropped silently. Please verify that the entry was accepted."
310 msgstr ""
311
312 msgid ""
313 "This configuration page allows you to configure a phone system (PBX) service "
314 "which permits making phone calls through multiple Google and SIP (like "
315 "Sipgate, SipSorcery, and Betamax) accounts and sharing them among many SIP "
316 "devices. Note that Google accounts, SIP accounts, and local user accounts "
317 "are configured in the \"Google Accounts\", \"SIP Accounts\", and \"User "
318 "Accounts\" sub-sections. You must add at least one User Account to this PBX, "
319 "and then configure a SIP device or softphone to use the account, in order to "
320 "make and receive calls with your Google/SIP accounts. Configuring multiple "
321 "users will allow you to make free calls between all users, and share the "
322 "configured Google and SIP accounts. If you have more than one Google and SIP "
323 "accounts set up, you should probably configure how calls to and from them "
324 "are routed in the \"Call Routing\" page. If you're interested in using your "
325 "own PBX from anywhere in the world, then visit the \"Remote Usage\" section "
326 "in the \"Advanced Settings\" page."
327 msgstr ""
328
329 msgid ""
330 "This is the name that the VoIP server will use to identify itself when "
331 "registering to VoIP (SIP) providers. Some providers require this to a "
332 "specific string matching a hardware SIP device."
333 msgstr ""
334
335 msgid ""
336 "This is where you indicate which Google/SIP accounts are used to call what "
337 "country/area codes, which users can use what SIP/Google accounts, how "
338 "incoming calls are routed, what numbers can get into this PBX with a "
339 "password, and what numbers are blacklisted."
340 msgstr ""
341
342 msgid ""
343 "This is where you set up your Google (Talk and Voice) Accounts, in order to "
344 "start using them for dialing and receiving calls (voice chat and real phone "
345 "calls). Please make at least one voice call using the Google Talk plugin "
346 "installable through the GMail interface, and then log out from your account "
347 "everywhere. Click \"Add\" to add as many accounts as you wish."
348 msgstr ""
349
350 msgid ""
351 "This is where you set up your SIP (VoIP) accounts ts like Sipgate, "
352 "SipSorcery, the popular Betamax providers, and any other providers with SIP "
353 "settings in order to start using them for dialing and receiving calls (SIP "
354 "uri and real phone calls). Click \"Add\" to add as many accounts as you wish."
355 msgstr ""
356
357 msgid ""
358 "This option should be set to \"Yes\" if you have a DID (real telephone "
359 "number) associated with this SIP account or want to receive SIP uri calls "
360 "through this provider."
361 msgstr ""
362
363 msgid ""
364 "This section contains settings that do not need to be changed under normal "
365 "circumstances. In addition, here you can configure your system for use with "
366 "remote SIP devices, and resolve call quality issues by enabling the "
367 "insertion of QoS rules."
368 msgstr ""
369
370 msgid ""
371 "Use (four to five digit) numeric user name if you are connecting normal "
372 "telephones with ATAs to this system (so they can dial user names)."
373 msgstr ""
374
375 msgid ""
376 "Use this account to make outgoing calls as configured in the \"Call Routing"
377 "\" section."
378 msgstr ""
379
380 msgid "Use this account to make outgoing calls."
381 msgstr ""
382
383 msgid "User Accounts"
384 msgstr ""
385
386 msgid "User Agent String"
387 msgstr ""
388
389 msgid "User Name"
390 msgstr ""
391
392 msgid "Uses providers enabled for outgoing calls"
393 msgstr ""
394
395 msgid ""
396 "When somebody starts voice chat with your GTalk account or calls the GVoice, "
397 "number (if you have Google Voice), the call will be forwarded to any users "
398 "that are online (registered using a SIP device or softphone) and permitted "
399 "to receive the call. If you have Google Voice, you must go to your GVoice "
400 "settings and forward calls to Google chat in order to actually receive calls "
401 "made to your GVoice number. If you have trouble receiving calls from GVoice, "
402 "experiment with the Call Screening option in your GVoice Settings. Finally, "
403 "make sure no other client is online with this account (browser in gmail, "
404 "mobile/desktop Google Talk App) as it may interfere."
405 msgstr ""
406
407 msgid ""
408 "When your password is saved, it disappears from this field and is not "
409 "displayed for your protection. The previously saved password will be changed "
410 "only when you enter a value different from the saved one."
411 msgstr ""
412
413 msgid "Yes"
414 msgstr ""
415
416 msgid ""
417 "You can enter your domain name, external IP address, or dynamic domain name "
418 "here Please keep in mind that if your IP address is dynamic and it changes "
419 "your configuration will become invalid. Hence, it's recommended to set up "
420 "Dynamic DNS in this case."
421 msgstr ""
422
423 msgid "You can specify a real name to show up in the Caller ID here."
424 msgstr ""
425
426 msgid ""
427 "You can use your SIP devices/softphones with this system from a remote "
428 "location as well, as long as your Internet Service Provider gives you a "
429 "public IP. You will be able to call other local users for free (e.g. other "
430 "Analog Telephone Adapters (ATAs)) and use your VoIP providers to make calls "
431 "as if you were local to the PBX. After configuring this tab, go back to "
432 "where users are configured and see the new Server and Port setting you need "
433 "to configure the remote SIP devices with. Please note that if this PBX is "
434 "not running on your router/gateway, you will need to configure port "
435 "forwarding (NAT) on your router/gateway. Please forward the ports below (SIP "
436 "port and RTP range) to the IP address of the device running this PBX."
437 msgstr ""
438
439 msgid ""
440 "Your PIN disappears when saved for your protection. It will be changed only "
441 "when you enter a value different from the saved one. Leaving the PIN empty "
442 "is possible, but please beware of the security implications."
443 msgstr ""
444
445 msgid ""
446 "Your password disappears when saved for your protection. It will be changed "
447 "only when you enter a value different from the saved one."
448 msgstr ""